This invention relates to digital convolution filters and particularly pertains to a digital signal receiver equipped with multiple simultaneous convolution processing of digital signals representative of sequential samples of an analog tone signal waveform.
Innovations in system design and fabrication techniques for circuit configurations have progressed to the extent that digital transmission for voice and data communication is in widespread use. One such use is in telephone systems where a substantial percentage of interoffice communication is by means of digital facilities. Such transmission enhances the quality, speed and privacy of communication while contributing to the retention of reasonable cost of telephone service. Digital transmission has proven particularly advantageous in switching systems where digital equipment is used for call processing in digital encoded form because the combination eliminates the need for many digital-to-analog and reverse conversions.
While substantial progress has been made in digital communications, a problem yet exists in present day systems where MF (multifrequency) signaling is utilized to convey data and instructions between originating and terminating switching offices in PCM (Pulse Code Modulation) form. The problem arises from the necessity to use complicated and expensive conversion equipments for restoring the digital signal to analog form so that the data and instructions can be interpreted. This technique is undesirable since it requires analog hardware which is bulky, costly and complex and frequently is not otherwise used in a digital switching system.
A need therefore exists in digital systems for facilities to digitally extract the signaling information from the PCM signal, and interpret it without the undesired additional process of first converting the PCM signal to analog.
MF signaling customarily utilizes combinations of any two of six tone signals having frequencies 700, 900, 1100, 1300, 1500 and 1700 Hertz. Each of the resultant signal combinations encoded in PCM format is a complex sequence of binary bits. Whether the MF signal originates in either an analog or a digital office, slight variations in amplitude and oscillator frequency, as well as background voice introduced before the tone signals are encoded into PCM format, can significantly change the sequence of bits. The structure and process used to detect the MF tones coded into these bits must therefore do so without full knowledge of the bits contained in the sequence. Only approximate information is known about the PCM encoded tone signals.
A number of designs for detecting digitally encoded MF tones in PCM systems is known.
U.S. Pat. No. 3,710,028 issued Jan. 9, 1973 to S. G. Pitroda discloses a technique for detecting MF tones in a PCM system based on a determination of the quantity of zero slope counts and peak detector counts for a specific period of time.
U.S. Pat. No. 3,824,471 issued July 16, 1974 to J. P. Mills uses Fourier spectrum analysis for detecting the presence of MF signals. The result of Fourier computations are accumulated and stored in floating point form and a determination is made whether the accumulated amplitude exceeds a certain percentage of that entire signal. The patent discloses an MF receiver employing the Fourier spectrum analysis. The magnitude of the signal is determined by summations of terms of the Fourier series expressed in terms of logs of sine and cosine expressions. The computations of 80 samples are accumulated for yielding an output result once every 10 milliseconds.
Another Mills U.S. Pat. No. 3,961,167 issued June 1, 1976 discloses a PCM tone receiver using statistical techniques. The probability that a certain tone was transmitted is based on a computer analysis of digital input samples and statistical determination that certain tones are present.
U.S. Pat. No. 3,872,290 issued Mar. 18, 1975 to A. W. Crooke discloses a digital filter using convolution of an input signal with a finite impulse function signal. The filter operates at an output sampling rate and partial results of the convolution are stored for a reduced number of required samples. As disclosed in the patent, finite impulse response filters can be realized by a direct application of the convolution equation: ##EQU1##
where h.sub.n are the impulse response coefficients of the filter, the x.sub.j represents the successive inputs to the filter, and y.sub.j represent the j.sup.th output from the filter.
U.S. Pat. No. 4,048,485 of Sept. 17, 1977 discloses a digital filter which includes a circular convolution device using the complex Mersenne transform to convert a sequence of input signal values into A.sub.n into another sequence A.sub.k in which: ##EQU2##
where P is a prime number and J is the square root of -1. The convolution is provided with an input for applying fast length data blocks made up of input samples appended with an equal number of zeros, circuits for recirculating and accumulating the data, a register for storing accumulated data and switches for selectively collecting the output of the storage to the inputs of an adder or subtractor. A product device multiplies term-by-term the output of the adder or subtractor with the complex Mersenne transform of filter coefficient sets appended with zeros and an inverse transform device for performing the inverse complex Mersenne transform in the multiplier output blocks of data.
An object of the present invention is to provide an improved digital convolution method and filter.